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gerrit-public.fairphone.software
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platform
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external
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webrtc
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3e33bfeb6d4a7b9e14ce856e849f5e0367689e09
3e33bfe
Fix some sign-compare warnings in webrtc/api.
by kjellander
· 8 years ago
839315b
Use the Chromium libfuzzer template instead of rolling our own.
by katrielc
· 8 years ago
1a20610
Fix buffer overflow in HMAC validation of STUN messages.
by katrielc
· 8 years ago
c853597
rtc::Buffer: Grow capacity by at least 1.5x to prevent quadratic behavior
by kwiberg
· 8 years ago
504f335
Roll chromium_revision 465d55d04e..6a85b3b953 (400622:400641)
by buildbot
· 8 years ago
ac62bd4
Rewrite CreateBlackFrame in webrtcvideoengine.
by nisse
· 8 years ago
44bf02f
Remove SdpAudioFormat's default constructor
by kwiberg
· 8 years ago
a7d88d3
Remove audio/video distinction for probe packets.
by Peter Boström
· 8 years ago
02343b9
Remove dead GYP target audio_device_module_java
by kjellander
· 8 years ago
442e6ee
Workaround java.gypi inclusion error in Chromium builds.
by kjellander
· 8 years ago
4c7f8ae
Cleanup MIPS specific link configuration
by kjellander
· 8 years ago
fc36a2d
Roll chromium_revision a21316a36e..465d55d04e (400620:400622)
by buildbot
· 8 years ago
ff702f5
Roll chromium_revision f66fe7e469..a21316a36e (400617:400620)
by buildbot
· 8 years ago
71687f3
Roll chromium_revision 5cdeb1b846..f66fe7e469 (400605:400617)
by buildbot
· 8 years ago
a9df50a
Roll chromium_revision 0962148116..5cdeb1b846 (400593:400605)
by buildbot
· 8 years ago
ce5a874
Improve encoding time calculation for Android HW encoder.
by glaznev
· 8 years ago
7508f3d
Roll chromium_revision 6c3ee789f0..0962148116 (400588:400593)
by buildbot
· 8 years ago
5023d41
GN: Update xmpp and p2p to match Chromium build
by kjellander
· 8 years ago
a935574
Roll chromium_revision e10a42e0d1..6c3ee789f0 (400570:400588)
by buildbot
· 8 years ago
cc5a582
Roll chromium_revision b078b9902f..e10a42e0d1 (400452:400570)
by buildbot
· 8 years ago
61a6946
Roll chromium_revision 3a1c71fcbd..b078b9902f (400409:400452)
by buildbot
· 8 years ago
f03a8d4
Unpack different wav files after each INIT event of the aecdump
by aluebs
· 8 years ago
863a826
Use |probe_cluster_id| to cluster packets.
by philipel
· 8 years ago
217fb66
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
by solenberg
· 8 years ago
3870001
Remove some dead code from VCMJitterBuffer.
by Tommi
· 8 years ago
57c21f9
Remove ViEEncoder::Pause / Start
by perkj
· 8 years ago
c13ded5
Move AudioCodingModuleImpl to anonymous namespace in audio_coding_module.cc
by kwiberg
· 8 years ago
434b85d
Roll chromium_revision ed2c9cb4cb..3a1c71fcbd (400384:400409)
by buildbot
· 8 years ago
ca6d5d1
Partial reland of Delete unused and almost unused frame-related methods. (patchset #1 id:1 of https://codereview.webrtc.org/2076113002/ )
by nisse
· 8 years ago
fd634c4
Reland of Re-enable UBsan on AGC.
by minyue
· 8 years ago
07ec26d
Fix crash parsing malformed rtp packet
by danilchap
· 8 years ago
9b99499
Added a builtin audio decoder factory to the default PeerConnectionFactory constructor.
by ossu
· 8 years ago
62379c8
Move Camera1 specific methods to Camera1Enumerator and create CameraEnumerator interface.
by sakal
· 8 years ago
72e735d
Revert of Delete unused and almost unused frame-related methods. (patchset #12 id:220001 of https://codereview.webrtc.org/2065733003/ )
by nisse
· 8 years ago
76270de
Delete unused and almost unused frame-related methods.
by nisse
· 8 years ago
0dbc8bf
Roll chromium_revision 1a73d11e65..ed2c9cb4cb (399420:400384)
by buildbot
· 8 years ago
e6c9e88
Android: Add Size class.
by sakal
· 8 years ago
50c4821
Fix missing resource file in webrtc_perf_tests.isolate
by kjellander
· 8 years ago
6af2e86
Refactor VideoDenoiser to work with I420Buffer, not VideoFrame.
by Niels Möller
· 8 years ago
6820889
Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420)
by kjellander
· 8 years ago
d1523ca
Fix header size check in PseudoTcp::parse().
by sergeyu
· 8 years ago
8e8222d
Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #4 id:290001 of https://codereview.webrtc.org/2071473002/ )
by tommi
· 8 years ago
e03f878
Reland of Split IncomingVideoStream into two implementations, with smoothing and without.
by tommi
· 8 years ago
4a0f7b5
- Remove use of VoERTP_RTCP::SetLocalSSRC() for receive streams; recreate them instead.
by solenberg
· 8 years ago
3abb764
Avoid unnecessary HW video encoder reconfiguration
by skvlad
· 8 years ago
9421853
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
by solenberg
· 8 years ago
e565a04
Revert of Fix crash parsing malformed rtp packet (patchset #1 id:1 of https://codereview.webrtc.org/2067793003/ )
by danilchap
· 8 years ago
5a45fe6
Fix crash parsing malformed rtp packet
by Danil Chapovalov
· 8 years ago
4867ca2
Revert of -enable UBsan on AGC. (patchset #1 id:1 of https://codereview.webrtc.org/2063643003/ )
by pbos
· 8 years ago
30a3a75
Fix buffer overflow parsing malformed rtp packet
by Danil Chapovalov
· 8 years ago
1642620
Performance fix for H264 RBSP parsing.
by Erik Språng
· 8 years ago
fc3a8ee
Delete unused code.
by Niels Möller
· 8 years ago
2d014be
Resolves issue with bad audio using BT headsets on iOS.
by henrika
· 8 years ago
5a9e7e0
Fix a few error prone lines on VideoCapturerAndroid.
by Sami Kalliomaki
· 8 years ago
b00dc38
Delete GetExecutablePath and related unused code.
by Niels Möller
· 8 years ago
342f740
NetEq: Ask AudioDecoder for sample rate instead of passing it as an argument
by kwiberg
· 8 years ago
347d351
AudioDecoder: Remove the default implementation of SampleRateHz
by kwiberg
· 8 years ago
371b43b
Changes synchronization offset perfomance tracking
by Danil Chapovalov
· 8 years ago
4f0dfbd
Change initial DTLS retransmission timer from 1 second to 50ms.
by Taylor Brandstetter
· 8 years ago
947c02d
Disable WebRtcVideoChannel2BaseTest.AddRemoveCapturer because it is flaky
by Alejandro Luebs
· 8 years ago
4c17abe
Add DesktopCapturer::Result::MAX_VALUE
by Sergey Ulanov
· 8 years ago
14461d4
Fixing flaky test: WebRtcSessionTest.TestPacketOptionsAndOnPacketSent
by deadbeef
· 8 years ago
a6219cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
by tommi
· 8 years ago
74290b9
New rtc dump analyzing tool in Python
by kjellander@webrtc.org
· 8 years ago
ceb9d0c
Audio decoder factory test: Ensure that g722's sample rate is 16 kHz, not 8 kHz
by kwiberg
· 8 years ago
6808419
iSAC decoder: Remove obsolete TODO
by kwiberg
· 8 years ago
edaa849
WebRtcVoiceCodecs: Eliminate some useless copying
by kwiberg
· 8 years ago
111744e
Added backwards compatible version of WebRtcMediaEngineFactory::Create.
by ossu
· 8 years ago
71ee44c
This cl:
by perkj
· 8 years ago
786f481
New misc scripts, header_usage.sh and author_line_count.sh.
by nisse
· 8 years ago
42883f8
Revert of Attempt to figure out what the issue is on the Win10 FYI build bot in content_browsertests. (patchset #1 id:1 of https://codereview.webrtc.org/2063313003/ )
by tommi
· 8 years ago
17c3cdd
Revert of Split IncomingVideoStream into two implementations, with smoothing and without. (patchset #23 id:430001 of https://codereview.webrtc.org/2035173002/ )
by tommi
· 8 years ago
37ad337
Remove EncodedFrameCallbackAdapter.
by sergeyu
· 8 years ago
204177f
Add RTCEventLog API to ObjC.
by tkchin
· 8 years ago
e110411
Attempt to figure out what the issue is on the Win10 FYI build bot in content_browsertests.
by tommi
· 8 years ago
2cc8baa
Adjust the amount of VP8 encoder threads for Android builds.
by Alex Glaznev
· 8 years ago
4deba9a
Add SigslotTester0 for testing signals without argument.
by honghaiz
· 8 years ago
8189b02
Configure VoE NACK through AudioReceiveStream::Config, for receive streams. Also minor refactoring of WVoE unit test.
by solenberg
· 8 years ago
184a3fd
Forward the SignalFirstPacketReceived to RtpReceiver.
by zhihuang
· 8 years ago
9a38cab
Voice Engine: Remove RED support
by kwiberg
· 8 years ago
5aaa9fa
Remove thread_checker in playout_delay_oracle
by isheriff
· 8 years ago
971cab0
Configure VoE NACK through AudioSendStream::Config, for send streams.
by solenberg
· 8 years ago
05b9803
Removed unused GetOutputVolume() and SetOutputVolume() from MediaEngineInterface.
by solenberg
· 8 years ago
8b06ec0
Change RTC_CHECK to RTC_CHECK_EQ for improved printout of GetLastError.
by tommi
· 8 years ago
6806136
Remove RED support from WebRtcVoiceEngine/MediaChannel
by kwiberg
· 8 years ago
b1963b4
Reland of Re-enable UBsan on AGC.
by minyue
· 8 years ago
dedfd28
Support for two audio codec lists down into WebRtcVoiceEngine.
by ossu
· 8 years ago
79ede03
Refactor VideoCapturerAndroid tests in WebRTC.
by sakal
· 8 years ago
1c7eef6
Split IncomingVideoStream into two implementations, with smoothing and without.
by tommi
· 8 years ago
e355069
Disable SctpDataMediaChannelTest.ReusesAStream.
by Peter Boström
· 8 years ago
0208322
GN: Add video_engine_tests
by Peter Boström
· 8 years ago
075af92
Initial asymmetric codec support in MediaSessionDescription
by ossu
· 8 years ago
87abc28
Add kwiberg@webrtc.org as root owner.
by solenberg
· 8 years ago
8660024
Remove webrtc_all target
by kjellander
· 8 years ago
7336225
Delete left-over files.
by nisse
· 8 years ago
1fc4810
Always on statistics for AndroidMediaEncoder.
by sakal
· 8 years ago
81d99b3
A missing path separator caused aecdump recordings
by peah
· 8 years ago
54f5a26
Report errors creating peer connection in AppRTC Demo Android.
by sakal
· 8 years ago
e9fc75e
Fixing SCTP verbose packet logging.
by deadbeef
· 8 years ago
dfe6937
Revert of Roll chromium_revision 7fa6701bc5..1a73d11e65 (398458:399420) (patchset #2 id:20001 of https://codereview.webrtc.org/2061723002/ )
by kjellander
· 8 years ago
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